1. Field of the Invention
The present invention is in the field of audio signal processing, more specifically in the field of audio dynamics processing utilizing variable gain elements to compress and expand audio frequency signals for noise reduction purposes.
2. State of the Art
In the 1970""s, companding noise reduction systemsxe2x80x94that is, audio systems employing complementary compression and expansion processesxe2x80x94became popular as a means to reduce the noise floor and increase the headroom of the analog magnetic tape recording process for higher fidelity, wider dynamic range recordings. By compressing (encoding) an input signal exhibiting a dynamic is range of 120 dB, for example, by a 2:1 ratio, the signal could then be passed through a transmission path, medium or xe2x80x9cnoisy channelxe2x80x9d having a limited dynamic range of 60 dB. Then, by complementarily expanding (decoding) the signal at the output of the transmission path or medium by a 1:2 ratio, the 120 dB dynamic range of the original signal would theoretically be preserved. This type of process was successfully applied to other xe2x80x9cnoisy channelxe2x80x9d transmission systems exhibiting limited dynamic range such as FM broadcasts and telecommunications. It should be understood that a transmission path, medium or channel is being used to refer to any type of audio processing system having a limited dynamic range.
In the last twenty years, the two most popular trade names associated with companding noise reduction systems have been dbx(copyright) and Dolby(copyright). A particularly important patent in this field by David E. Blackmer, U.S. Pat. No. 3,789,143, granted Jan. 29, 1974, describes a method of dbx-type noise reduction wherein compression and expansion are complementary in time response. Principles in this patent form the basis of complementary noise reduction systems used by dbx(copyright), Dolby(copyright), and others. The typical design of these systems, as illustrated in the aforementioned Blackmer patent, consists of two separate circuits-an encoder circuit and a decoder circuit. Each of these circuits, in their most simple form, includes a main audio path consisting of a variable gain element having an audio input port and an audio output port, and a detector path or xe2x80x9csidechainxe2x80x9d having a circuit that detects the audio signal level and creates a control signal. Typically, there are other circuits in the sidechain which are xe2x80x9cdownstreamxe2x80x9d from the detector, whose function is to shape or process the control signalxe2x80x94this processed control signal being used to control the gain of the variable gain element. More complex noise reduction schemes may include additional circuits in the sidechain that pre-process the audio xe2x80x9cupstreamxe2x80x9d from the level detector.
In each of these well-known systems, there exists two sidechains-one for the encoder and one for the decoder. Typically, the encoder uses a feedback circuit topology wherein the input signal to the encoder sidechain is the audio output signal of the encoder. Conversely, the typical circuit topology of the decoder is feedforward wherein the input signal to the decoder sidechain is the audio input signal of the decoder. These sidechains are substantially identical other than the fact that they create control voltages which cause opposite reactions in their respective variable gain elements. That is, when one variable gain element is adding gain to the audio signal passing through it, the other is complementarily attenuating the audio signal by the same amount.
The main advantage of this feedback encoder and feedforward decoder arrangement is its complementary nature because the input signal that the decoder sidechain xe2x80x9cseesxe2x80x9d is substantially the same input signal that the encoder sidechain sees having only the noisy channel between the inputs of the two sidechains. Of course, the noisy channel will add some noise and distortion and may exhibit frequency response anomalies which would alter the xe2x80x9ccleanxe2x80x9d signal that the encoder is acting upon, thus presenting an altered signal to the decoder, but with properly designed sidechain circuitry, the sidechain detectors react appropriately to the signal. Thus, with substantially identical encoder and decoder sidechains acting upon substantially the same signal, the decoder operates in a substantially complementary manner to the encoder.
A person unfamiliar with noise reduction systems might naturally question the need for having two separate sidechains in the system, especially when the circuitry is substantially identical in each, and each detector is intended to react to substantially the same signal. The answer is, that for traditional uses of noise reduction it is not possible to share sidechain circuitry without adding unnecessary complexity to the system. For example, in analog tape recording, encoding occurs during the recording process making use of the encoder sidechain. Because playback can occur in a different location than where the recording took place, the decoder must have its own sidechain electronics in the playback system. A similar need for the separate decoder sidechain is obviously needed for noise reduction on radio broadcasts as well. One way around this is to record the decode signal onto tape in parallel with the audio signal or, for broadcasts, to transmit the decode signal along with the audio. Both methods would add unnecessary complexity to the encoder as well as require additional bandwidth or an extra channel in the transmission path, neither of which are desirable nor required to obtain reasonable performance.
However, as is well known to those skilled in the art, one of the biggest problems with these types of noise reduction systems is that they require components of the decoder sidechain to be closely matched with corresponding components of the encoder sidechain to avoid frequency and gain errors. Also, the operating levels of the system comprising the xe2x80x9cnoisy channelxe2x80x9d must be calibrated to the same operating levels to which the compander is calibrated to avoid mistracking and gain errors. If close matching is not maintained, this xe2x80x9csemi-complementaryxe2x80x9d processing may produce very audible and unpleasant artifacts. This is not easy to do in manufacturing and components may drift over time and through aging and temperature cycling. Other mistracking errors occur when the noisy channel does not behave in a predictable manner. For instance, analog tape may have xe2x80x9cdropoutsxe2x80x9d in level and various brands and types of tape, or even various levels of quality within one type of tape, will cause errors and mistracking due to inconsistent frequency response and other non-linearities.
Another problem associated with noise reduction systems, due to the minimum compression and expansion ratios-typically 2:1 or greater-required to achieve satisfactory levels of noise reduction and increased headroom through the limited dynamic range channel, is that the noise level is audibly modulated by the level of the audio signal, a phenomenon known as xe2x80x9cbreathing.xe2x80x9d Other problems include dynamic distortion of low frequency waveforms if the level detectors are too fast and thus track these waveforms, and distortion of fast, high-level transients if the level detectors are not fast enough to react to the audio. Lastly, the sheer number of components in good noise reduction systems makes them costly to implement on a per-channel basis.
Turning the focus to the present invention, the inventors considered the problem of how to achieve a wider dynamic range from a system that many audio professionals never really considered to be the xe2x80x9cweak linkxe2x80x9d in the audio chain. For years, in most systems, either analog tape or the broadcast channel was the limiting factor in overall system dynamic range even when employing a noise reduction system such as dbx(copyright). The electronic noise from solid state devices such as equalizers was a minor factor and was often completely ignored. But as digital recording brought increased dynamic range to the audio signal chain, people began paying more attention to the electronic noise contributed by other equipment and began looking more closely at the equipment""s printed specifications. In the specific case of an audio equalizer, it is not difficult for a manufacturer of graphic equalizers to design an equalizer with dynamic range specifications exceeding 106 dB, where the noise floor is 90 dB below the nominal signal level and the clip point is 16 dB above the nominal signal level. This is more than 10 dB greater dynamic range than typical 16-bit digital performance.
This 106 dB dynamic range specification for a graphic equalizer would lead the inexperienced, as well as many veteran audio professionals, to believe that their equalizer exhibits more-than-adequate noise performance for digital recording. But the truly experienced professional knows that when the audio signal is equalized, this is done at the expense of increased noise levels, not simply due to boosting the noise xe2x80x9cupstreamxe2x80x9d from the equalizer, but also due to self-noise generated within the equalizer itself which is most apparent in critical recording or when equalizing a live concert system where the noise floor is boosted to concert levels. The experienced professional knows, as well as the designer of the equalizer, that this specification is measured with all frequency gain controls set in the 0 dB or xe2x80x9cflatxe2x80x9d position which is only a partially useful specification because it does not represent the real noise performance of the equalizer in actual use. The designer of the equalizer (hopefully) knows that the 90 dB signal-to-noise ratio specification is destroyed as soon as any frequency band is boosted. Likewise, this self-generated equalizer noise is increased when any frequency band is cut-a fact of which many users of equalizers employing active circuitry are not aware. The 90 dB signal-to-noise ratio can easily be degraded to only 60 to 70 dB with radical equalization or degraded to 75 to 85 dB with only moderate equalization.
Self-generated equalizer noise is due to the electronic noise of the active filter or filters in the signal path of the particular frequency band or bands which are adjusted away from the 0 dB position. Also, any noise magnetically or capacitively coupled into the equalizer filter circuits will become apparent as frequency bands are adjusted away from 0 dB. As is well known by designers of equalizers, when a particular frequency band is in the flat position, the audio signal and noise generated by or coupled into that respective active filter circuit is nulled out by either the filter summing circuits or by shorting the filter output to ground, thus eliminating its noise contribution to the overall noise floor. For obvious reasons, manufacturers are hesitant to publish the whole truth of what happens to the noise floor with any amount of equalization. Also, to be fair, it would be difficult for the audio industry to standardize on a specification for measurement of noise that could be applied equitably to all equalizers given the variety of designs with their optimum and worst case settings varying from one design to another.
The result of using an equalizer in a real world situation is that it often becomes the weak link in the system when considering noise performance. So, although many audio professionals have not really considered this, an equalizer is just another example of a limited dynamic range system. With modern audio systems, higher fidelity performance is required from each component in the audio chain. By fitting an equalizer with a properly designed companding noise reduction system, a substantial improvement in the noise performance of the equalizer can be realized.
Thus, it would be an improvement over the state of the art to improve the performance of applicable weak links in the audio chain using a novel noise reduction topology and sidechain circuit for a class of limited dynamic range systems. Although existing noise reduction systems could be used to partially accomplish this basic purpose, they fail to solve the real problem. Specifically, an improved noise reduction system should be designed for limited dynamic range systems, such as graphic or parametric equalizers, which exhibit minimal or no time delay through the transmission channel. The improved system should work best when there is no delay, but should also be designed to handle minimal delays such as phase shifts through filter circuits while maintaining audible integrity.
It would be a further improvement to make the system adaptable to systems with longer than minimal delays, a digital delay for example, by delaying the decode signal by the same delay inherent in the transmission channel.
It is an object of the present invention to design a novel noise reduction system that does not exhibit the aforementioned problems of previous systems for use with limited dynamic range systems exhibiting minimal or no delay.
Specifically, a principal objective of the present invention is to provide a noise reduction system that is substantially audibly transparent for use in modern audio systems which exhibit wider dynamic range than, say, analog tape, thus requiring better sonic characteristics than existing noise reduction systems. Thus, to meet this objective, the present invention needs to minimize audible noise modulation or xe2x80x9cbreathingxe2x80x9d; to minimize dynamic distortion of low frequency signals due to the level detector tracking these waveforms; to minimize distortion of signals due to phase shift in the transmission channel; to minimize distortion of fast, high-level transients due to the attack time of the level detector; and to have virtually no effect on the audio signal if the level of the signal stays within a reasonable range around the nominal signal level.
Another objective of the present invention is to design a noise reduction system that avoids the inherent problems of component matching between the encoder and the decoder due to component tolerances, component aging and temperature drifts.
Another objective of the present invention is to make the noise reduction circuit free of level matching adjustments over the life of the product, where settings of noise reduction parameters determined by electrical component choices would yield substantially repeatable results in manufacturing and stable functionality in field use.
Another objective of the present invention is to eliminate, where possible, any errors due to non-linearities associated with the noisy channel itself which would result in the decoder not functioning in an adequately complementary manner to the encoder.
A final objective of the present invention is to design a noise reduction circuit that is relatively inexpensive so that it can be included in the circuitry of any system which could benefit from the use of this noise reduction without incurring excessive added cost that would unduly burden the retail price of the system.
One of the novel features of the invention which makes accomplishing these objectives possible is a unique noise reduction circuit topology wherein the encoder and decoder share the same feedforward sidechain, thus generating a single control voltage derived from the characteristics of only one signal for simultaneously controlling their respective variable gain elements. This one signal is the audio input signal to the encoder making the sidechain, by definition, a feedforward type. This topology, as opposed to the aforementioned methods utilizing two sidechains whose level detectors do not react to precisely the same signal, means that it is possible to meet the objective of avoiding the errors associated with component aging, drifting, and matching between the encoder and decoder by default as their sidechains are one and the same.
It is also possible to meet the objective of avoiding the need to match any levels between the encoder and the decoder sidechains because there is only one sidechain. Also, because this sidechain xe2x80x9clooks atxe2x80x9d only one audio signal to derive its variable gain element control voltage, it is possible to at least partially meet the objective that any errors due to non-linearities associated with the noisy channel itself are eliminated, because the decoder does not react to the altered audio signal that has passed through the noisy channel.
The presently preferred embodiment of the invention is substantially audibly transparent, thereby minimizing audible noise modulation, dynamic distortion of low frequency signals, distortion of signals due to phase shift in the transmission channel, and distortion of fast, and high-level transients. These objections are accomplished while there is virtually no effect at nominal signal levels as a result of the benefits gained by the feedforward circuit topology, by use of the preferred RMS method of level detection, by the design of the novel timing circuit, and by wise choice of noise reduction parameters, namely low and high threshold points. The design of the novel timing circuit also further satisfies the objective of minimizing errors due to non-linearities associated with the noisy channel itself which would result in the decoder not functioning in an adequately complementary manner to the encoder.
By employing a feedforward sidechain circuit topology, the distortion of low frequency signals is minimized by decreasing audio gain modulation due to the ripple on the control voltage controlling the variable gain element. A feedforward sidechain circuit topology exhibits less ripple on the control voltage than an equivalent feedback sidechain circuit topology designed with the same attack and release time constants to achieve an equivalent amount of signal compression. A thorough explanation of this fact can be found in Application Note 101Axe2x80x94The Mathematics of Log-Based Dynamic Processors, Rev. Sep. 28, 1995, produced by THAT Corporation, 734 Forest Street, Marlborough, Mass. 01752.
Additionally, by employing the preferred RMS method of level detection, in wide use and well known to those skilled in the art and aptly described in U.S. Pat. No. 3,681,618, issued to David E. Blackmer, it is again possible to gain the benefit that this method exhibits low rectification ripple which further reduces distortion of low frequency signals due to gain modulation of the audio signal by the ripple on the control voltage.
Also, it is well known that the human ear hears loudness in proportion to the RMS energy and thus the RMS detector responds appropriately to audio signals as compared to peak or averaging detectors. Additionally, this RMS detector exhibits the desirable property that it naturally operates at various speeds, reacting faster to higher slew rate signals and slower to slower moving signals. Further, this detector is of the xe2x80x9cdecilinearxe2x80x9d type; that is, its output is linear in volts versus a logarithmic change of input signal voltage. The output level from this Blackmer RMS detector follows a control law which is specified in mV per dB.
With a decilinear detector, it is easy to add, subtract, and process control voltages in the xe2x80x9clog domainxe2x80x9d to thereby derive the proper variable gain element control signal, especially when using a preferred type of variable gain element which follows the same decilinear control law. One such variable gain element is the widely used dbx(copyright) uPC1252HA2 VCA, which is a monolithic integrated circuit manufactured by NEC Electronics Inc., available from THAT Corporation, Marlborough, Mass. under their 215X series part numbers. Those skilled in the art will know how to configure this dbx(copyright) VCA for compression and expansion functions.
The novel timing circuit is designed to be a non-linear low-pass filter such that it has multiple speeds of operation or time constants. Means are provided so that it reacts slowly to low frequency signals to avoid the distortion associated with tracking these waveforms. Other means are provided so that it reacts quickly to high-level transients avoiding overload distortion associated with attack times that are too slow to reduce the gain of the encoder. Further means are provided so that it releases quickly when the audio signal decays quickly to avoid audible noise modulation or xe2x80x9cbreathing.xe2x80x9d Lastly, means are provided to reduce dynamic distortion of low frequency signals without compromising the fast release time needed to avoid noise modulation which also minimizes distortion due to phase shift inherent in the transmission channel.
Through wise choice of thresholds, it is further possible to maximize audio transparency. As is well-known by those skilled in the art, noise modulation occurs in the output vs. input gain region where the audio signal is above the low threshold and below the high threshold. Below the low threshold, the attenuation of noise is constant and independent of the audio signal level, while above the high threshold, no attenuation of noise is occurring. In the region between the two thresholds, attenuation of the noise is a function of the audio signal level related by the compander ratio. A 2:1 ratio means that if the audio signal level changes by 1 dB, the noise changes by 2 dB. This change in the noise floor modulated by the audio signal is audible, which is the main reason existing noise reduction schemes limit the compander ratio to a low value such as 2:1 or, at most, 3:1. By setting the low compander threshold of the preferred embodiment significantly above the noise floor, it is possible to minimize the audibility of the noise modulation because the audio signal level is high enough to mask it. By setting the high threshold of the preferred embodiment significantly below the nominal audio signal level, the noise reduction is not doing anything the majority of the time with nominal signal levels. This is perfectly acceptable and desirable because nominal signal levels are very high compared to the noise floor and mask any noise present, and the variable gain elements exhibit optimum audio performance at unity gain when they are not amplifying or attenuating.
In a simple compander noise reduction system, there are four level-related output vs. input gain parameters: low threshold, high threshold, ratio, and amount of noise reduction with three degrees of freedom from which to choose, the fourth being determined by the choice of the other three. By choosing a low threshold point significantly above the noise floor, say xe2x88x9255 dBu, and a high threshold point significantly below the nominal signal level, say xe2x88x9225 dBu, and by choosing a reasonable amount of noise reduction in dB, say 24 dB, the ratio is automatically fixed at 5:1. This high ratio value would cause unacceptably audible artifacts in existing noise reduction systems, but due to the novel topology with a single feedforward sidechain employing simultaneous encode and decode, and due to the novel timing circuit and prudent choice of thresholds allowing plenty of xe2x80x9cslopxe2x80x9d above the noise floor and below nominal signal levels, it is thus possible to achieve the objective of audible transparency. Artifacts are virtually nonexistent and are certainly not audible in normal use.
Furthermore, the desired output vs. input gain parameters can be set by choosing fixed values of electrical components without being overly concerned with tight tolerances or component drifts with age or temperature because audible variability in the preferred embodiment is no longer related to component matching. Also, with xe2x80x9cslopxe2x80x9d designed into the low and high threshold parameters, and with the ability to tolerate high compansion ratios, functional variability is hardly a consideration with the noise reduction system of the preferred embodiment because even ten percent changes in component values do not affect its subjective operational characteristics. This robust design of the preferred embodiment allows the luxury of not having to trim any component values in manufacturing, and the end-user does not have to calibrate it in the field. Thus, the objective of the choice of components yielding substantially repeatable results in manufacturing and stable functionality in field use is met.
Another objective that the noise reduction system is to be relatively inexpensive is met because the implementation of the sidechain circuitry is much simpler than that of any good noise reduction system, where one sidechain is completely eliminated with its associated level detector and control voltage processing circuits as well as the level matching circuitry included in the sidechains of existing systems.
These and other objects, features, advantages and alternative aspects of the present invention will become apparent to those skilled in the art from a consideration of the following detailed description taken in combination with the accompanying drawings.